The Internet, like so many other high tech developments, grew from research originally performed by the United States Department of Defense. In the 1960s, the military had accumulated a large collection of incompatible computer networks. Computers on these different networks could not communicate with other computers across their network boundaries.
In the 1960s, the Defense Department wanted to develop a communication system that would permit communication between these different computer networks. Recognizing that a single, centralized communication system would be vulnerable to attacks or sabotage, the Defense Department required that the communication system be decentralized with no critical services concentrated in vulnerable failure points. In order to achieve this goal, the Defense Department established a decentralized standard communication protocol for communication between their computer networks.
A few years later, the National Science Foundation (NSF) wanted to facilitate communication between incompatible network computers at various research institutions across the country. The NSF adopted the Defense Department's protocol for communication, and this combination of research computer networks would eventually evolve into the Internet.
Internet Protocols
The Defense Department's communication protocol governing data transmission between different networks was called the Internet Protocol (IP) standard. The IP standard has been widely adopted for the transmission of discrete information packets across network boundaries. In fact, the IP standard is the standard protocol governing communications between computers and networks on the Internet.
The IP standard identifies the types of services to be provided to users and specifies the mechanisms needed to support these services. The IP standard also specifies the upper and lower system interfaces, defines the services to be provided on these interfaces, and outlines the execution environment for services needed in the system.
In a typical Internet-based communication scenario, data is transmitted from an originating communication device on a first network across a transmission medium to a destination communication device on a second network. After receipt at the second network, the packet is routed through the network to a destination communication device using standard addressing and routing protocols. Because of the standard protocols in Internet communications, the IP protocol on the destination communication device decodes the transmitted information into the original information transmitted by the originating device.
The IP-Based Mobility System
The Internet protocols were originally developed with an assumption that Internet users would be connected to a single, fixed network. With the advent of cellular wireless communication systems using mobile communication devices, the movement of Internet users within a network and across network boundaries has become common. Because of this highly mobile Internet usage, the implicit design assumption of the Internet protocols (e.g. a fixed user location) is violated by the mobility of the user.
In an IP-based mobile communication system, the mobile communication device (e.g. cellular phone, pager, computer, etc.) can be called a mobile node or mobile station. Typically, a mobile station maintains connectivity to its home network while operating on a visited network. The mobile station will always be associated with its home network for IP addressing purposes and will have information routed to it by routers located on the home and visited networks.
Packet-Based Communication Systems
In Internet Protocol (IP) networks, the communication process is very different from prior conventional telecommunication systems. In an IP network communication, there is no open switched connection established between the caller and recipient devices. The information being transmitted between the caller and recipient devices is broken into packets of data, and each packet of data is transmitted to the recipient device in pieces. The data packets individually contain routing information to direct each packet to the recipient device. These packets are then reassembled into a coherent stream of data at the recipient device.
Code Division Multiple Access (CDMA) is an evolving third generation communication system standard for wireless communication systems that can transmit multimedia services using the packet-based Internet protocol. These CDMA mobile communication systems support multimedia telecommunication services delivering voice such as VoIP (Voice over IP) and data, to include pictures, video communications, and other multimedia information over mobile wireless connections. These types of communications are typically time-sensitive and require high data rate transfers with inherent delays minimized as much as possible.
As the capability of the various communication standards have improved, there has been an increasing need for high-speed transmissions and increased user capacity. A new CDMA packet air interface has been developed that offers improvements over earlier CDMA systems by implementing high-speed shared-traffic packet data channels on the forward air-link connection. Recent developments include CDMA-based 1xEV systems operating at 1.25 MHz. The 1.25 MHz carrier delivers high data rates and increased voice capacity. 1xEV is a two-phase strategy. One phase is designated 1xEV-DO, which handles data only. The 1xEV-DO standard provides user with peak data rates of 3.0 Mbits/s. The other phase is 1xEV-DV, for data and voice. Other standards are evolving that also make use of the shared packet channel and multiplex packet communication for high-speed data and voice communication.
In the CDMA standard, Mobile Nodes, or Access Terminals (AT), roam within and across cellular communication sites. Each of the sites, or cells, possesses one or more transceivers coupled to a Base Transceiver Station (BTS) onto the communication network. The BTSs are in turn coupled to an Access Network (AN), also known as a Radio Network (RN). As an AT travels across cellular borders, its physical connection to the BTS keeps on changing. An AT can be physically located anywhere on the network or sub-network, and its routing address data will change and require updating on other nodes. Wireless IP networks handle the mobile nature of AT with hand-off procedures designed to update the communication network and sub-network with the location of the mobile node for packet routing purposes. The latency period in these hand-offs can be prohibitively high. Call (or packet data session) setup times can also be excessive as communication pathways across the access network (whether wireline or wireless) are established before transmitting session specific data (e.g., a SIP Invite message) needed to establish a delay sensitive application—such as VoIP, PTT (Push To Talk) and VT (Video Telephony) etc.—session.
A new method of call (or packet data session) signaling (for example, SIP signaling) to setup a real-time application session (such as a VoIP, PTT or VT session) in 1xEV-DO can significantly reduce the call setup time. Call (or packet data session) setup time is an important performance indicator for applications like Push-to-Talk (PTT), Voice over IP (VoIP), and Video Telephony (VT). Call (or packet data session) setup is a critical aspect of delay sensitive application (such as VoIP, PTT, and VT etc.) functionality, since call (or packet data session) setup delays can negate the value of the function. Long pauses in walkie-talkie (or voice call) performance adversely impact the versatility to using the function and lead to customer dissatisfaction. Service providers are quite aware of this concern and are insistent that implementing infrastructure minimizes call (or packet data session) setup delays.
Applications using wireless technology for data delivery have to explicitly reserve network resources before data can be exchanged to meet unique resource requirements for the content being transmitted. Resource reservation has to be done by the originating and terminating parties for satisfactory communication since various media content have specific requirements in terms of delay and jitter tolerance. The content can be classified as delay sensitive, rate sensitive, time sensitive, or some other quality adversely impacting the communication. The method for reserving and allocating network resources can significantly effect call (or packet data session) setup time. The extra messaging involved in setting up resources prior to initiating media flow increases call (or packet data session) setup times that often fall outside the limits of tolerance.
In the invention, the RNC learns the nature of the content from the requests reserved by the originator of media, and sets aside resources a priori for the terminating end without waiting for a request (from the terminating AT). This earlier reservation or allocation of resources at the terminating end without a request provides immense savings in call (or packet data session) setup times, which is critical for successful deployment of essential services such as PTT, VoIP, and VT, etc. Since resource availability is known during the early stages of call (or a packet data session) setup, the certainty of the stability of established calls (or packet data sessions) can be assured with a higher degree of success.
Prior art approaches have not been successful. An optimistic approach to call (or packet data session) setup has been proposed as a predictive scheme. This method involves the delivery of a call (or packet data session) announcement as a general broadcast along with an indication that resources have been allocated beforehand. This allotment is made without a localized view of resource availability or knowledge of which paging sector the AT may respond on. Once the location of the AT is known through its response, subsequent resource allocation attempts by the network on a specific channel within the location sector may fail, causing the call (or packet data session) attempt to abort. Furthermore, this predictive approach is not scalable because it can only be used over a limited number of paging sectors and requires the RN to become content aware. The range limitation of paging sectors covered decreases the first page success rate, destroying the benefits of predictive resource allocation. Content awareness is also computing resource intensive and may adversely impact performance indicators across all applications. There is a need for a new method of resource allocation that reduces latency and speeds up call (or packet data session) setup in time sensitive applications such as PTT, VoIP, and VT, which have stringent constraints on call (or packet data session) set up delays.